Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/pc/rtpsender.h

Index: webrtc/pc/rtpsender.h
diff --git a/webrtc/api/rtpsender.h b/webrtc/pc/rtpsender.h
similarity index 98%
copy from webrtc/api/rtpsender.h
copy to webrtc/pc/rtpsender.h
index 6c0592148bb06d264bdb84b3bdff51b262b03e08..ed244b416a7226e77cc5d340c8557c4c77effb40 100644
--- a/webrtc/api/rtpsender.h
+++ b/webrtc/pc/rtpsender.h
@@ -12,19 +12,19 @@
 // An RtpSender associates a MediaStreamTrackInterface with an underlying
 // transport (provided by AudioProviderInterface/VideoProviderInterface)
 
-#ifndef WEBRTC_API_RTPSENDER_H_
-#define WEBRTC_API_RTPSENDER_H_
+#ifndef WEBRTC_PC_RTPSENDER_H_
+#define WEBRTC_PC_RTPSENDER_H_
 
 #include <memory>
 #include <string>
 
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/rtpsenderinterface.h"
-#include "webrtc/api/statscollector.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/pc/channel.h"
+#include "webrtc/pc/statscollector.h"
 
 namespace webrtc {
 
@@ -232,4 +232,4 @@ class VideoRtpSender : public ObserverInterface,
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_API_RTPSENDER_H_
+#endif  // WEBRTC_PC_RTPSENDER_H_